Chapter 6: Multimedia Networking

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Chapter 6 Multimedia Networking

• Our goals
• principles network, application-level support
  for multimedia
• different forms of network multimedia,
  requirements
• making the best of best effort service
• mechanisms for providing QoS
• specific streaming protocols
• architectures for QoS

• Overview
• multimedia applications and requirements
• making the best of todays best effort service
• scheduling and policing mechanisms
• next generation Internet
• Intserv
• RSVP
• Diffserv

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Multimedia, Quality of Service What is it?
Multimedia applications network audio and video
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Multimedia Performance Requirements

• Requirement deliver data in timely manner
• interactive multimedia short end-end delay
• e.g., IP telephony, teleconf., virtual worlds,
  DIS
• excessive delay impairs human interaction
• streaming (non-interactive) multimedia
• data must arrive in time for smooth playout
• late arriving data introduces gaps in rendered
  audio/video
• reliability 100 reliability not always required
  

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MM Networking Applications

• Classes of MM applications
• 1) Streaming stored audio and video
• 2) Streaming live audio and video
• 3) Real-time interactive audio and video

• Fundamental characteristics
• Typically delay sensitive
• end-to-end delay
• delay jitter
• But loss tolerant infrequent losses cause minor
  glitches
• Antithesis of data, which are loss intolerant but
  delay tolerant

Jitter is the variability of packet delays
within the same packet stream
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Streaming Stored Multimedia

• Streaming
• media stored at source
• transmitted to client
• streaming client playout begins before all data
  has arrived

• timing constraint for still-to-be transmitted
  data in time for playout

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Streaming Stored Multimedia What is it?
Cumulative data
time
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Streaming Multimedia - Interactivity

• Types of interactivity
• none like broadcast radio, TV
• initial startup delays of lt 10 secs OK
• VCR-functionality client can pause, rewind, FF
• 1-2 sec until command effect OK

• timing constraint for still-to-be transmitted
  data in time for playout

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Streaming Live Multimedia

• Examples
• Internet radio talk show
• Live sporting event (e.g., soccer game)
• Streaming
• playback buffer
• playback can lag tens of seconds after
  transmission
• still have timing constraint
• Interactivity
• fast forward impossible
• rewind, pause possible!

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Interactive, Real-Time Multimedia

• applications IP telephony, video conference,
  distributed interactive worlds

• end-end delay requirements
• audio lt 150 msec good, lt 400 msec OK
• includes application-level (packetization) and
  network delays
• higher delays noticeable, impair interactivity
• session initialization
• how does callee advertise its IP address, port
  number, encoding algorithms?

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Multimedia Over Todays Internet

• TCP/UDP/IP best-effort service
• no guarantees on delay, loss

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How should the Internet evolve to better support
multimedia?

• Integrated services philosophy
• Fundamental changes in Internet so that apps can
  reserve end-to-end bandwidth
• Requires new, complex software in hosts routers
• Laissez-faire
• no major changes
• more bandwidth when needed
• content distribution, application-layer
  mechanisms
• application layer

• Differentiated services philosophy
• Fewer changes to Internet infrastructure, yet
  provide 1st and 2nd class service.

Whats your opinion?
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Streaming Stored Multimedia

• Application-level streaming techniques for making
  the best out of best effort service
• client side buffering
• use of UDP versus TCP
• multiple encodings of multimedia

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Internet multimedia simplest approach

• audio or video stored in file
• files transferred as HTTP object
• received in entirety at client
• then passed to player

• audio, video not streamed
• no, pipelining, long delays until playout!

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Internet multimedia streaming approach

• browser GETs metafile
• browser launches player, passing metafile
• player contacts server
• server streams audio/video to player

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Streaming from a streaming server

• This architecture allows for non-HTTP protocol
  between server and media player
• Can also use UDP instead of TCP.

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Streaming Multimedia Client Buffering
constant bit rate video transmission
Cumulative data
time

• Client-side buffering, playout delay compensate
  for network-added delay, delay jitter

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Streaming Multimedia Client Buffering
constant drain rate, d
variable fill rate, x(t)
buffered video

• Client-side buffering, playout delay compensate
  for network-added delay, delay jitter

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Streaming Multimedia UDP or TCP?

• UDP
• server sends at rate appropriate for client
  (oblivious to network congestion !)
• often send rate encoding rate constant rate
• then, fill rate constant rate - packet loss
• short playout delay (2-5 seconds) to compensate
  for network delay jitter
• error recover time permitting
• TCP
• send at maximum possible rate under TCP
• congestion loss fill rate fluctuates
• larger playout delay smooth TCP delivery rate
• HTTP/TCP passes more easily through firewalls

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Streaming Multimedia client rate(s)
1.5 Mbps encoding
28.8 Kbps encoding

• Q how to handle different client receive rate
  capabilities?
• 28.8 Kbps dialup
• 100Mbps Ethernet

A server stores, transmits multiple copies of
video, encoded at different rates
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Real-time interactive applications

• Lets look at a PC-2-PC Internet phone example
  in detail.

• PC-2-PC phone
• instant messaging services are providing this
• PC-2-phone
• Dialpad
• Net2phone
• videoconference with Webcams

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Interactive Multimedia Internet Phone

• Introduce Internet Phone by way of an example
• speakers audio alternating talk spurts, silent
  periods.
• 64 kbps during talk spurt
• pkts generated only during talk spurts
• 20 msec chunks at 8 Kbytes/sec 160 bytes data
• application-layer header added to each chunk
• chunkheader encapsulated into UDP segment
• application sends UDP segment into socket every
  20 msec during talk spurt

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Internet Phone Packet Loss and Delay

• network loss IP datagram lost due to network
  congestion (router buffer overflow)
• delay loss IP datagram arrives too late for
  playout at receiver
• delays processing, queueing in network
  end-system (sender, receiver) delays
• typical maximum tolerable delay 400 ms
• loss tolerance depending on voice encoding,
  losses concealed, packet loss rates between 1
  and 10 can be tolerated.

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Delay Jitter
constant bit
rate transmission
Cumulative data
time

• Consider the end-to-end delays of two consecutive
  packets difference can be more or less than 20
  msec

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Internet Phone Fixed Playout Delay

• Receiver attempts to playout each chunk exactly q
  msecs after chunk was generated.
• chunk has time stamp t play out chunk at tq .
• chunk arrives after tq data arrives too late
  for playout, data lost
• Tradeoff for q
• large q less packet loss
• small q better interactive experience

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Fixed Playout Delay

• Sender generates packets every 20 msec during
  talk spurt.
• First packet received at time r
• First playout schedule begins at p
• Second playout schedule begins at p

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Recovery From Packet Loss

• loss packet never arrives or arrives too late
• real-time constraints little (no) time for
  retransmissions!
• What to do?
• Forward Error Correction (FEC) add error
  correction bits (recall 2-dimensional parity)
• add redundant chunk made up of exclusive OR of n
  chunks
• redundancy (overhead) is 1/n
• can reconstruct if at most one lost chunk
• Interleaving spread loss evenly over received
  data to minimize impact of loss

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FEC - Piggybacking Lower Quality Stream

• FEC Scheme
• piggyback lower quality stream
• send lower resolution audio stream as the
  redundant information
• Whenever there is non-consecutive loss, the
  receiver can conceal the loss
• Can also append (n-1)st and (n-2)nd low-bit rate
  chunk

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Interleaving

• Interleaving Scheme
• no redundancy needed
• chunks are brokenup into smaller units
• for example, four 5 msec units per chunk
• packet contains small units from different chunks

• if packet is lost, still have most of every chunk
• has no redundancy overhead
• but adds to playout delay

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Summary Internet Multimedia bag of tricks

• use UDP to avoid TCP congestion control (delays)
  for time-sensitive traffic
• client-side adaptive playout delay to compensate
  for delay
• server side matches stream bandwidth to available
  client-to-server path bandwidth
• chose among pre-encoded stream rates
• dynamic server encoding rate
• error recovery (on top of UDP)
• FEC, interleaving
• retransmissions, time permitting
• conceal errors repeat nearby data

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Improving QOS in IP Networks

• Thus far making the best of best effort
• Future next generation Internet with QoS
  guarantees
• RSVP signaling for resource reservations
• Differentiated Services differential guarantees
• Integrated Services firm guarantees
• simple model for sharing and congestion
  studies

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Principles for QOS Guarantees

• Example 1Mbps IP phone, FTP share 1.5 Mbps
  link.
• bursts of FTP can congest router, cause audio
  loss
• want to give priority to audio over FTP

Principle 1
packet marking needed for router to distinguish
between different classes and new router policy
to treat packets accordingly
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Principles for QOS Guarantees (more)

• what if applications misbehave (audio sends
  higher than declared rate)
• policing force source adherence to bandwidth
  allocations
• marking and policing at network edge
• similar to ATM UNI (User Network Interface)

Principle 2
provide protection (isolation) for one class from
others
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Principles for QOS Guarantees (more)

• Allocating fixed (non-sharable) bandwidth to
  flow inefficient use of bandwidth if flows
  doesnt use its allocation

Principle 3
While providing isolation, it is desirable to use
resources as efficiently as possible
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Principles for QOS Guarantees (more)

• Basic fact of life can not support traffic
  demands beyond link capacity

Principle 4
Call Admission flow declares its needs, network
may block call (e.g., busy signal) if it cannot
meet needs
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Summary of QoS Principles
Lets next look at mechanisms for achieving this
.
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Scheduling And Policing Mechanisms

• scheduling choose next packet to send on link
• FIFO (first in first out) scheduling send in
  order of arrival to queue
• real-world example?
• discard policy if packet arrives to full queue
  who to discard?
• Tail drop drop arriving packet
• priority drop/remove on priority basis
• random drop/remove randomly

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Scheduling Policies more

• Priority scheduling transmit highest priority
  queued packet
• multiple classes, with different priorities
• class may depend on marking or other header info,
  e.g. IP source/dest, port numbers, etc..
• Real world example?

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Scheduling Policies still more

• round robin scheduling
• multiple classes
• cyclically scan class queues, serving one from
  each class (if available)
• real world example?

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Scheduling Policies still more

• Weighted Fair Queuing
• generalized Round Robin
• each class gets weighted amount of service in
  each cycle
• real-world example?

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Policing Mechanisms

• Goal limit traffic to not exceed declared
  parameters
• Three common-used criteria
• (Long term) Average Rate how many pkts can be
  sent per unit time (in the long run)
• crucial question what is the interval length
  100 packets per sec or 6000 packets per min have
  same average!
• Peak Rate e.g., 6000 pkts per min. (ppm) avg.
  1500 ppm peak rate
• (Max.) Burst Size max. number of pkts sent
  consecutively (with no intervening idle)

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Policing Mechanisms

• Token Bucket limit input to specified Burst Size
  and Average Rate.
• bucket can hold b tokens
• tokens generated at rate r token/sec unless
  bucket full
• over interval of length t number of packets
  admitted less than or equal to (r t b).

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Policing Mechanisms (more)

• token bucket, WFQ combine to provide guaranteed
  upper bound on delay, i.e., QoS guarantee!

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IETF Integrated Services

• architecture for providing QOS guarantees in IP
  networks for individual application sessions
• resource reservation routers maintain state info
  (a la VC) of allocated resources, QoS reqs
• admit/deny new call setup requests

Question can newly arriving flow be admitted
with performance guarantees while not violated
QoS guarantees made to already admitted flows?
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Intserv QoS guarantee scenario

• Resource reservation
• call setup, signaling (RSVP)
• traffic, QoS declaration
• per-element admission control

request/ reply
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Call Admission

• Arriving session must
• declare its QoS requirement
• R-spec defines the QoS being requested
• characterize traffic it will send into network
• T-spec defines traffic characteristics
• signaling protocol needed to carry R-spec and
  T-spec to routers (where reservation is required)
• RSVP

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Intserv QoS Service models rfc2211, rfc 2212

• Guaranteed service
• worst case traffic arrival leaky-bucket-policed
  source
• simple (mathematically provable) bound on delay
  Parekh 1992, Cruz 1988

• Controlled load service
• "a quality of service closely approximating the
  QoS that same flow would receive from an unloaded
  network element."

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IETF Differentiated Services

• Concerns with Intserv
• Scalability signaling, maintaining per-flow
  router state difficult with large number of
  flows
• Flexible Service Models Intserv has only two
  classes. Also want qualitative service classes
• behaves like a wire
• relative service distinction Platinum, Gold,
  Silver
• Diffserv approach
• simple functions in network core, relatively
  complex functions at edge routers (or hosts)
• Dot define define service classes, provide
  functional components to build service classes

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Diffserv Architecture
Edge router - per-flow traffic management -
marks packets as in-profile and out-profile
Core router - per class traffic management -
buffering and scheduling based on marking at
edge - preference given to in-profile packets -
Assured Forwarding
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Edge-router Packet Marking

• profile pre-negotiated rate A, bucket size B
• packet marking at edge based on per-flow profile

User packets
Possible usage of marking

• class-based marking packets of different classes
  marked differently
• intra-class marking conforming portion of flow
  marked differently than non-conforming one

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Classification and Conditioning

• Packet is marked in the Type of Service (TOS) in
  IPv4, and Traffic Class in IPv6
• 6 bits used for Differentiated Service Code Point
  (DSCP) and determine PHB that the packet will
  receive
• 2 bits are currently unused

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Classification and Conditioning

• may be desirable to limit traffic injection rate
  of some class
• user declares traffic profile (eg, rate, burst
  size)
• traffic metered, shaped if non-conforming

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Forwarding (PHB)

• PHB result in a different observable (measurable)
  forwarding performance behavior
• PHB does not specify what mechanisms to use to
  ensure required PHB performance behavior
• Examples
• Class A gets x of outgoing link bandwidth over
  time intervals of a specified length
• Class A packets leave first before packets from
  class B

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Forwarding (PHB)

• PHBs being developed
• Expedited Forwarding pkt departure rate of a
  class equals or exceeds specified rate
• logical link with a minimum guaranteed rate
• Assured Forwarding 4 classes of traffic
• each guaranteed minimum amount of bandwidth
• each with three drop preference partitions

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Multimedia Networking Summary

• multimedia applications and requirements
• making the best of todays best effort service
• scheduling and policing mechanisms
• next generation Internet
• Intserv, RSVP, Diffserv